|Summary:||ASTERISK-15147: MixMonitor stops audio on native bridging|
|Date Opened:||2009-11-16 14:41:56.000-0600||Date Closed:||2011-06-07 14:00:58|
|Description:||If I run mixmonitor before running dial, no audio is heard on the calling party's end. When I comment out the mixmonitor line, everything works perfectly. |
****** ADDITIONAL INFORMATION ******
My extensions.conf looks as follows:
exten => 1234,1,Answer
;exten => 1234,n,MixMonitor(/var/lib/asterisk/recordings/call.wav)
exten => 1234,n,Dial(SIP/somenumber,90,gL(3600000))
;exten => 1234,n,StopMixMonitor
exten => 1234,n,hangup()
with those lines commented out, I hear audio perfectly, as soon as I uncomment them, I hear silence. The audio file gets recorded but only records my side of the conversation. The asterisk interface looks as follows:
-- Executing [####@default:1] NoOp("SIP/some_ip-0925a940", "") in new stack
-- Executing [####@default:2] MixMonitor("SIP/some_ip-0925a940", "/var/lib/asterisk/recordings/call.wav") in new stack
-- Executing [####@default:3] Dial("SIP/some_ip-0925a940", "SIP/someNumber,90,gL(3600000)") in new stack
-- Setting call duration limit to 3600.000 seconds.
== Using SIP RTP CoS mark 5
-- Called someNumber
== Begin MixMonitor Recording SIP/some_ip-0925a940
-- SIP/someNumber-09248ae0 is ringing
-- SIP/someNumber-09248ae0 answered SIP/some_ip-0925a940
== Spawn extension (default, someNumber, 4) exited non-zero on 'SIP/some_ip-0925a940'
== MixMonitor close filestream
== End MixMonitor Recording SIP/someIP-0925a940
I am doing this all over SIP protocol on version 126.96.36.199
|Comments:||By: Leif Madsen (lmadsen) 2010-01-13 13:39:15.000-0600|
Since I haven't had a chance to test and confirm this, can you let me know if this is still an issue on the latest version of Asterisk? If so then I'll go ahead and try to get some testing done to confirm. Thanks!
By: Leif Madsen (lmadsen) 2010-03-23 09:59:49
Suspended due to lack of response from reporter.